Voicing index controls for CELP speech coding

ABSTRACT

An approach for improving quality of speech synthesized using analysis-by-synthesis (ABS) coders is presented. An unstable perceptual quality in analysis-by-synthesis type speech coding (e.g. CELP) may occur because the periodicity degree in a voiced speech signal may vary significantly for different segments of the voiced speech. Thus, the present invention uses a voicing index, which may indicate the periodicity degree of the speech signal, to control and improve ABS type speech coding. The voicing index may be used to improve the quality stability by controlling encoder and/or decoder in: fixed-codebook short-term enhancement including the spectrum tilt; perceptual weighting filter; sub-fixed codebook determination; LPC interpolation; fixed-codebook pitch enhancement; post-pitch enhancement; noise injection into the high-frequency band at decoder; LTP Sinc window; signal decomposition, etc.

RELATED APPLICATIONS

[0001] The present application claims the benefit of U.S. provisionalapplication serial No. 60/455,435, filed Mar. 15, 2003, which is herebyfully incorporated by reference in the present application.

[0002] The following co-pending and commonly assigned U.S. patentapplications have been filed on the same day as this application, andare incorporated by reference in their entirety:

[0003] U.S. patent application Ser. No. ______, “SIGNAL DECOMPOSITION OFVOICED SPEECH FOR CELP SPEECH CODING,” Attorney Docket Number: 0160112.

[0004] U.S. patent application Ser. No. ______, “SIMPLE NOISESUPPRESSION MODEL,” Attorney Docket Number: 0160114.

[0005] U.S. patent application Ser. No. ______, “ADAPTIVE CORRELATIONWINDOW FOR OPEN-LOOP PITCH,” Attorney Docket Number: 0160115.

[0006] U.S. patent application Ser. No. ______, “RECOVERING AN ERASEDVOICE FRAME WITH TIME WARPING,” Attorney Docket Number: 0160116.

BACKGROUND OF THE INVENTION

[0007] 1. Field of the Invention

[0008] The present invention relates generally to speech coding and,more particularly, to Code Excited Linear Prediction (CELP) speechcoding.

[0009] 2. Related Art

[0010] Generally, a speech signal can be band-limited to about 10 kHzwithout affecting its perception. However, in telecommunications, thespeech signal bandwidth is usually limited much more severely. It isknown that the telephone network limits the bandwidth of the speechsignal to between 300 Hz to 3400 Hz, which is known as the “narrowband”.Such band-limitation results in the characteristic sound of telephonespeech. Both the lower limit at 300 Hz and the upper limit at 3400 Hzaffect the speech quality.

[0011] In most digital speech coders, the speech signal is sampled at 8kHz, resulting in a maximum signal bandwidth of 4 kHz. In practice,however, the signal is usually band-limited to about 3600 Hz at thehigh-end. At the low-end, the cut-off frequency is usually between 50 Hzand 200 Hz. The narrowband speech signal, which requires a samplingfrequency of 8 kb/s, provides a speech quality referred to as tollquality. Although this toll quality is sufficient for telephonecommunications, for emerging applications such as teleconferencing,multimedia services and high-definition television, an improved qualityis necessary.

[0012] The communications quality can be improved for such applicationsby increasing the bandwidth. For example, by increasing the samplingfrequency to 16 kHz, a wider bandwidth, ranging from 50 Hz to about 7000Hz can be accommodated, which is referred to as the “wideband”.Extending the lower frequency range to 50 Hz increases naturalness,presence and comfort. At the other end of the spectrum, extending thehigher frequency range to 7000 Hz increases intelligibility and makes iteasier to differentiate between fricative sounds.

[0013] Digitally, speech is synthesized by a well-known approach knownas Analysis-By-Synthesis (ABS). Analysis-By-Synthesis is also referredto as closed-loop approach or waveform-matching approach. It offersrelatively better speech coding quality than other approaches for mediumto high bit rates. A known ABS approach is the so-called Code ExcitedLinear Prediction (CELP). In CELP coding, speech is synthesized by usingencoded excitation information to excite a linear predictive coding(LPC) filter. The output of the LPC filter is compared against thevoiced speech and used to adjust the filter parameters in a closed loopsense until the best parameters based upon the least error is found. Oneof the facts influencing CELP coding is that voicing degree cansignificantly vary for different voiced speech segments thus causing anunstable perceptual quality in the speech coding.

[0014] The present invention addresses the above analysis-by-synthesisvoiced speech issue.

SUMMARY OF THE INVENTION

[0015] In accordance with the purpose of the present invention asbroadly described herein, there is provided systems and methods forimproving quality of synthesized speech by using a voicing index tocontrol the speech coding process.

[0016] According to one embodiment of the present invention, a voicingindex is used to control and improve ABS type speech coding, whichindicates the periodicity degree of the speech signal. The periodicitydegree can significantly vary for different voiced speech segments, andthis variation causes an unstable perceptual quality inanalysis-by-synthesis type speech coding, such as CELP.

[0017] The voicing index can be used to improve the quality stability bycontrolling encoder and/or decoder, for example, in the following areas:(a) fixed-codebook short-term enhancement including the spectrum tilt,(b) perceptual weighting filter, (c) sub-fixed codebook determination,(d) LPC interpolation, (e) fixed-codebook pitch enhancement, (f)post-pitch enhancement, (g) noise injection into the high-frequency bandat decoder, (h) LTP Sinc window, (i) signal decomposition, etc. In oneembodiment for CELP speech coding, the voicing index may be based on anormalized pitch correlation.

[0018] These and other aspects of the present invention will becomeapparent with further reference to the drawings and specification, whichfollow. It is intended that all such additional systems, methods,features and advantages be included within this description, be withinthe scope of the present invention, and be protected by the accompanyingclaims.

BRIEF DESCRIPTION OF DRAWINGS

[0019]FIG. 1 is an illustration of the frequency domain characteristicsof a sample speech signal.

[0020]FIG. 2 is an illustration of a voicing index classificationavailable to both the encoder and the decoder.

[0021]FIG. 3 is an illustration of a basic CELP coding block diagram.

[0022]FIG. 4 is an illustration of a CELP coding process with anadditional adaptive weighting filter for speech enhancement inaccordance with an embodiment of the present invention.

[0023]FIG. 5 is an illustration of a decoder implementation with postfilter configuration in accordance with an embodiment of the presentinvention.

[0024]FIG. 6 is an illustration of a CELP coding block diagram withseveral sub-codebooks.

[0025]FIG. 7A is an illustration of sampling for creation of a Sincwindow.

[0026]FIG. 7B is an illustration of a Sinc window.

DETAILED DESCRIPTION

[0027] The present application may be described herein in terms offunctional block components and various processing steps. It should beappreciated that such functional blocks may be realized by any number ofhardware components and/or software components configured to perform thespecified functions. For example, the present application may employvarious integrated circuit components, e.g., memory elements, digitalsignal processing elements, transmitters, receivers, tone detectors,tone generators, logic elements, and the like, which may carry out avariety of functions under the control of one or more microprocessors orother control devices. Further, it should be noted that the presentapplication may employ any number of conventional techniques for datatransmission, signaling, signal processing and conditioning, tonegeneration and detection and the like. Such general techniques that maybe known to those skilled in the art are not described in detail herein.

[0028] Voicing index is traditionally one of the important indexes sentto the decoder for Harmonic speech coding. The voicing index generallyrepresents the degree of periodicity and/or periodic harmonic bandboundary of voiced speech. Voicing index is traditionally not used inCELP coding systems. However, embodiments of the present invention usethe voicing index to provide control and improve the quality ofsynthesized speech in a CELP or other analysis-by-synthesis type coder.

[0029]FIG. 1 is an illustration of the frequency domain characteristicsof a sample speech signal. In this illustration, the spectrum domain inthe wideband extends from slightly above 0 Hz to around 7 kHz. Althoughthe highest possible frequency in the spectrum ends at 8 kHz (i.e.Nyquist folding frequency) for a speech signal sampled at 16 kHz, thisillustration shows that the energy is almost zero in the area between7.0 kHz to 8 kHz. It should be apparent to those of skill in the artsthat the ranges of signals used herein are for illustration purposesonly and that the principles expressed herein are applicable to othersignal bands.

[0030] As illustrated in FIG. 1, the speech signal is quite harmonic atlower frequencies, but at higher frequencies the speech signal does notremain as harmonic because the probability of having noisy speech signalincreases as the frequency increases. For instance, in this illustrationthe speech signal exhibits traits of becoming noisy at the higherfrequencies, e.g., above 5.0 kHz. This noisy signal makes waveformmatching at higher frequencies very difficult. Thus, techniques like ABScoding (e.g. CELP) becomes unreliable if high quality speech is desired.For example, in a CELP coder, the synthesizer is designed to match theoriginal speech signal by minimizing the error between the originalspeech and the synthesized speech. A noisy signal is unpredictable thusmaking error minimization very difficult.

[0031] Given the above problem, embodiments of the present invention usea voicing index which is sent to the decoder, from the encoder, toimprove the quality of speech synthesized by an ABS type speech coder,e.g., CELP coder.

[0032] The voicing index, which is transmitted by the encoder to thedecoder, may represent the periodicity of the voiced speech or theharmonic structure of the signal. In another example embodiment, thevoicing index may be represented by three bits thus providing up toeight classes of speech signal. For instance, FIG. 2 is an illustrationof a voicing index classification available to both the encoder and thedecoder. In this illustration, index 0 (i.e. “000”) may indicatebackground noise, index 1 (i.e. “001”) may indicate noise-like orunvoiced speech signal, index 2 (i.e. “010”) may indicate irregularvoiced signal such as voiced signal during onset, and indices 3-7 (i.e.“011” to “111”) could each indicate the periodicity of the speechsignals. For instance, index 3 (“011”) may represent the least periodicsignal and index 7 (“111”) may indicate the most periodic signal.

[0033] The voicing index information can be transmitted by the encoderas part of each encoded frame. In other words, each frame may includethe voicing index bits (e.g. three bits), which indicate the periodicitydegree of that particular frame. In one embodiment, the voicing indexfor CELP may be based on a normalized pitch correlation parameter, Rp,and may be derived from the following equation: 10 log(1−Rp)², where−1.0<Rp<1.0.

[0034] In one example, the voicing index may be used for fixed codebookshort-term enhancement, including the spectrum tilt. FIG. 3 is anillustration of a basic CELP coding block diagram. As illustrated, theCELP coding block 300 comprises the Fixed Codebook 301, gain block 302,Pitch filter block 303, and LPC filter 304. CELP coding block 300further comprises comparison block 306, Weighting Filter block 320, andMean Squared Error (MSE) computation block 308.

[0035] The basic idea behind CELP coding is that Input Speech 307 iscompared against the synthesized output 305 to generate error 309, whichis the mean squared error. The computation continues in a closed loopsense with selection of a new coding parameters until error 309 isminimal.

[0036] On the receiving side, the decoder synthesizes the speech usingsimilar blocks 301-304 (see FIG. 5). Thus, the encoder passesinformation to the decoder as needed to select the proper codebookentry, gain, and filters, . . . , etc . . .

[0037] In a CELP speech coding system, when the speech signal is moreperiodic, the pitch filter (e.g. 303) contribution is heavier than thefixed codebook (e.g. 301) contribution. As a result, an embodiment ofthe present invention may use the voicing index to place more focus inthe high frequency region by implementing an adaptive high pass filter,which is controlled by the value of the voicing index. An architecturesuch as the one shown in FIG. 4 may be implemented. For instance,Adaptive Filter 310 could be an adaptive filter emphasizing the power inthe high frequency region. In the illustration, the weighting filter 420may also be an adaptive filter for improving the CELP coding process.

[0038] On the decoder side, the voicing index may be used to select theappropriate Post Filter 520 parameters. FIG. 5 is an illustration of thedecoder implementation with post filter configuration. In one or moreembodiments, Post Filter 520 may have several configurations saved in atable, which may be selectable using information in the voicing index.

[0039] In another example, the voicing index may be used in conjunctionwith the perceptual weighting filter of CELP. The perceptual weightingfilter may be represented by Adaptive filter 420 of FIG. 4, for example.As is well known, waveform matching minimizes the error in the mostimportant portion (i.e. the high energy portion) of the speech signaland ignores low energy area by performing a mean squared errorminimization. Embodiments of the present invention use an adaptiveweighting process to enhance the low energy area. For instance, thevoicing index may be used to define the aggressiveness of the weightingfilter 420 depending on the periodicity degree of the frame.

[0040] In yet another embodiment, as illustrated in FIG. 6, the voicingindex may be used to determine the sub-fixed codebook. There arepossibly several sub-codebooks for the fixed codebook, for example, onesub-codebook 601 with less pulses but higher position resolution, onesub-codebook 602 with more pulses but lower position resolutions, and anoise sub-codebook 603. Therefore, if the voicing index indicates anoisy signal, then the sub-codebook 602 or noisy sub-codebook 603 can beused; if the voicing index does not indicate a noisy signal, then one ofthe sub-codebooks (e.g. 601 or 602) may be used depending on the degreeof periodicity of the given frame. Note that the gain block (codebook)302 may also be applied individually to each sub-codebook in one or moreembodiments.

[0041] Further, the voicing index may be used in conjunction with theLPC interpolation. For example, during linear interpolation, theprevious LPC is equally important as the current LPC if the location ofthe interpolated LPC is at the middle between the previous one and thecurrent one. Thus, if the voicing index, for example, indicates that theprevious frame was unvoiced and the present frame is voiced, then duringthe LPC interpolation, the LPC interpolation algorithm may favor thecurrent frame more than the previous

[0042] The voicing index may also be used for fixed codebook pitchenhancement. Typically, the previous pitch gain is used to perform pitchenhancement. However, the voicing index provides information relating tothe current frame and, thus, could be a better indicator than theprevious pitch gain information. The magnitude of the pitch enhancementmay be determined based on the voicing index. In other words, the moreperiodic the frame (based on the voicing index value), the higher themagnitude of the enhancement. For example, the voicing index may be usedin conjunction with the U.S. patent application Ser. No. 09/365,444,filed Aug. 2, 1999, specification of which is incorporated herein byreference, to determine the magnitude of the enhancements in thebi-directional pitch enhancement system defined therein.

[0043] As a further example, the voicing index may be used in place ofpitch gain for post pitch enhancement. This is advantageous, since, asdiscussed above, the voicing index may be derived from a normalizedpitch correlation value, i.e. Rp, which is typically between 0.0 and1.0; however, pitch gain may exceed 1.0 and can adversely affect thepost pitch enhancement process.

[0044] As another example, the voicing index may also be used todetermine the amount of noise that should be injected in the highfrequency band at the decoder side. This embodiment may be used when theinput speech is decomposed into a voiced portion and a noise portion asdiscussed in pending U.S. patent application Ser. No. ______, filedconcurrently herewith, entitled “SIGNAL DECOMPOSITION OF VOICED SPEECHFOR CELP SPEECH CODING”, specification of which is incorporated hereinby reference.

[0045] The voicing index may also be used to control modification of theSinc window. The Sinc window is used to generate an adaptive codebookcontribution vector, i.e. LTP excitation vector, with fractional pitchlag for CELP coding. In wideband speech coding, it is known that strongharmonics appear in the low frequency area of the band and the noisysignals appear in the high frequency area.

[0046] Long-term prediction or LTP produces the harmonics by taking aprevious excitation and copying it to a current subframe according tothe pitch period. It should be noted that if a pure copy of the previousexcitation is made, then the harmonic is replicated all the way to theend spectrum in the frequency domain. However, that would not be anaccurate representation of a true voice signal and especially not inwideband speech coding.

[0047] In one embodiment, for wideband speech signal when the previoussignal is used to represent the current signal, an adaptive low passfilter is applied to the Sinc interpolation window, since there is ahigh probability of noise in high frequency area.

[0048] In CELP coding, the fixed codebook contributes to coding of thenoisy or irregular portion of the speech signal, and a pitch adaptivecodebook contributes to the voice or regular portion of the speechsignal. The adaptive codebook contribution is generated using a Sincwindow, which is used due to the fact that the pitch lag can befractional. If the pitch lag were an integer, one excitation signalcould be copied to the next; however, because the pitch lag isfractional, straight copying of the previous excitation signal would notwork. After the Sinc window is modified, the straight copying would notwork even for integer pitch lag. In order to generate pitchcontribution, several samples are taken, as shown in FIG. 7A, which areweighted and then added together, where the weights for the samples iscalled the Sinc window, which originally has a symmetric shape, as shownin FIG. 7B. The shape in practice depends on the fractional portion ofthe pitch lag and the adaptive lowpass filter applied to the Sincwindow. Application of the Sinc window is similar to convolution orfiltering, but the Sinc window is a non-causal filter. In therepresentation shown below, a window signal w(n) is convoluted with thesignal s(n) in the time domain, which is an equivalent representation tospectrum of the window W(w) multiplied by the spectrum of the signalS(w) in the frequency domain:

U _(ACB)(n)=w(n)*s(n)←→W(w)S(w).

[0049] According to the above representation, low passing of the Sincwindow is equivalent to low passing the final adaptive codebookcontribution (UACB (n)) or excitation signal; however, low passing ofthe Sinc window is advantageous due to the fact that the Sinc window isshorter than the excitation. Thus, it is easier to modify the Sincwindow than the excitation; further more, the filtering of the Sincwindow can be pre-calculated and memorized.

[0050] In one embodiment of the present invention, the voicing index maybe used to provide information to control modification of the low passfilter for the Sinc window. For instance, the voicing index may provideinformation as to whether the harmonic structure is strong or weak. Ifthe harmonic structure is strong, then a weak low pass filter is appliedto the Sinc window, and if the harmonic structure is weak, then a stronglow pass filter is applied to the Sinc window.

[0051] Although the above embodiments of the present application aredescribed with reference to wideband speech signals, the presentinvention is equally applicable to narrowband speech signals.

[0052] The methods and systems presented above may reside in software,hardware, or firmware on the device, which can be implemented on amicroprocessor, digital signal processor, application specific IC, orfield programmable gate array (“FPGA”), or any combination thereof,without departing from the spirit of the invention. Furthermore, thepresent invention may be embodied in other specific forms withoutdeparting from its spirit or essential characteristics. The describedembodiments are to be considered in all respects only as illustrativeand not restrictive.

What is claimed is:
 1. A method of improving synthesized speech qualitycomprising: obtaining an input speech signal; coding said input speechusing a Code Excited Linear Prediction coder to generate code parametersfor synthesis of said input speech; and using a voicing indexrepresenting a characteristic of said input speech in enhancing saidsynthesis of said input speech.
 2. The method of claim 1, wherein saidcharacteristic of said input speech is periodicity of said input speech.3. The method of claim 1, wherein said enhancing said synthesis of saidinput speech is by controlling an adaptive highpass filter with saidvoicing index to enhance high frequency region during said coding. 4.The method of claim 1, wherein said enhancing said synthesis of saidinput speech is by controlling an adaptive perceptual weighting filterin said Code Excited Linear Prediction coder with said voicing index. 5.The method of claim 1, wherein said enhancing said synthesis of saidinput speech is by controlling an adaptive Sinc window used in said CodeExcited Linear Prediction coder for pitch contribution with said voicingindex.
 6. The method of claim 1, wherein said enhancing said synthesisof said input speech is by controlling spectrum tilt of said inputspeech by short-term enhancement of a fixed-codebook of said CodeExcited Linear Prediction coder with said voicing index.
 7. The methodof claim 1, wherein said enhancing said synthesis of said input speechis by controlling a perceptual weighting filter of said Code ExcitedLinear Prediction coder with said voicing index.
 8. The method of claim1, wherein said enhancing said synthesis of said input speech is bycontrolling a linear prediction coder of said Code Excited LinearPrediction coder with said voicing index.
 9. The method of claim 1,wherein said enhancing said synthesis of said input speech is bycontrolling a pitch enhancement fixed-codebook of said Code ExcitedLinear Prediction coder with said voicing index.
 10. The method of claim1, wherein said enhancing said synthesis of said input speech is bycontrolling post pitch enhancement of said Code Excited LinearPrediction coder with said voicing index.
 11. The method of claim 1,wherein said voicing index selects at least one sub-codebook from aplurality of sub-codebooks of said Code Excited Linear Prediction coderbased on said characteristic of said input speech signal.
 12. A methodof improving synthesized speech quality comprising: obtaining codeparameters of an input speech signal; obtaining a voicing index for usein enhancing synthesis of said input speech signal from said codeparameters; and processing said code parameters through a Code ExcitedLinear Prediction coder using information provided by said voicing indexto generate a synthesized version of said input speech signal.
 13. Themethod of claim 12, wherein said voicing index provides periodicity ofsaid input speech signal.
 14. The method of claim 12, wherein saidvoicing index provides characteristics of an adaptive highpass filterused to enhance high frequency region of said excitation duringgeneration of said code parameters for said input speech.
 15. The methodof claim 12, wherein said voicing index provides characteristics of anadaptive perceptual weighting filter used to enhance perceptual qualityof said input speech during generation of said code parameters for saidinput speech.
 16. The method of claim 12, wherein said voicing indexprovides characteristics of an adaptive Sinc window for pitchcontribution used to enhance perceptual quality of said input speechduring generation of said code parameters for said input speech.
 17. Themethod of claim 12, wherein said enhancing synthesis of said inputspeech is by controlling spectrum tilt of said input speech byshort-term enhancement of a fixed-codebook of said Code Excited LinearPrediction coder with said voicing index.
 18. The method of claim 12,wherein said enhancing of said synthesis of said input speech is bycontrolling a linear prediction coder filter of said Code Excited LinearPrediction coder with said voicing index.
 19. The method of claim 12,wherein said enhancing of said synthesis of said input speech is bycontrolling a pitch enhancement fixed-codebook of said Code ExcitedLinear Prediction coder with said voicing index.
 20. The method of claim12, wherein said enhancing said synthesis of said input speech is bycontrolling post pitch enhancement of said Code Excited LinearPrediction coder with said voicing index.
 21. The method of claim 12,wherein said voicing index selects at least one sub-codebook from aplurality of sub-codebooks of said Code Excited Linear Prediction coderbased on said characteristic of said input speech signal.
 22. Anapparatus for improving synthesized speech quality comprising: an inputspeech signal; a Code Excited Linear Prediction coder for coding saidinput speech signal to generate code parameters for synthesis of saidinput speech; and a voicing index having a characteristic of said inputspeech for use in enhancing said synthesis of said input speech.
 23. Theapparatus of claim 22, wherein said characteristic of said input speechis periodicity of said input speech.
 24. The apparatus of claim 22,wherein said characteristic of said input speech is a characteristic ofan adaptive highpass filter used to enhance high frequency region ofsaid excitation during said coding.
 25. The apparatus of claim 22,wherein said characteristic of said input speech is a characteristic ofan adaptive perceptual weighting filter used in said Code Excited LinearPrediction coder.
 26. The apparatus of claim 22, wherein saidcharacteristic of said input speech is a characteristic of an adaptiveSinc window used in said Code Excited Linear Prediction coder.
 27. Theapparatus of claim 22, wherein said voicing index selects at least onesub-codebook from a plurality of sub-codebooks of said Code ExcitedLinear Prediction coder based on said characteristic of said inputspeech signal.
 28. An apparatus for improving synthesized speech qualitycomprising: a set of code parameters of an input speech signal; avoicing index for use in enhancing synthesis of said input speech signalfrom said code parameters; and a Code Excited Linear Prediction coderusing said code parameters and information provided by said voicingindex to generate a synthesized version of said input speech signal. 29.The apparatus of claim 28, wherein said voicing index providesperiodicity of said input speech signal.
 30. The apparatus of claim 28,wherein said voicing index provides characteristics of a highpass filterused to enhance high frequency region of said excitation duringgeneration of said code parameters for said input speech.
 31. Theapparatus of claim 28, wherein said voicing index providescharacteristics of an adaptive perceptual weighting filter used toenhance perceptual quality of said input speech during generation ofsaid code parameters for said input speech.
 32. The apparatus of claim28, wherein said voicing index provides characteristics of an adaptiveSinc window used to enhance perceptual quality of said input speechduring generation of said code parameters for said input speech.
 33. Theapparatus of claim 28, wherein said voicing index selects at least onesub-codebook from a plurality of sub-codebooks of said Code ExcitedLinear Prediction coder based on characteristics of said input speechsignal.
 34. A method of improving synthesized speech quality comprising:generating a plurality of frames from an input speech signal; codingeach frame of said plurality of frames using a Code Excited LinearPrediction coder to generate code parameters for synthesis of said eachframe of said input speech; and transmitting a voicing index having aplurality of bits indicative of a classification of said each frame ofsaid input speech.
 35. The method of claim 34, wherein said plurality ofbits are three bits.
 36. The method of claim 34, wherein saidclassification is indicative of periodicity of said input speech signal.37. The method of claim 34, wherein said classification is indicative ofan irregular voiced speech signal.
 38. The method of claim 34, whereinsaid classification is indicative of a periodic index.
 39. The method ofclaim 38, wherein said periodic index ranges from low periodic index tohigh periodic index.
 40. A method of improving synthesized speechquality comprising: receiving a frame of an input speech signal, saidframe having a plurality of code parameters and a voicing index, whereinsaid voicing index comprises a plurality of bits; determining aclassification for said frame of said input speech signal from saidplurality of bits of said voicing index; and decoding said frame using aCode Excited Linear Prediction coder based on said classification tosynthesize said input speech.
 41. The method of claim 40, wherein saidplurality of bits are three bits.
 42. The method of claim 40, whereinsaid classification is indicative of a noisy speech signal.
 43. Themethod of claim 40, wherein said classification is indicative of anirregular voiced speech signal.
 44. The method of claim 40, wherein saidclassification is indicative of a periodic index.
 45. The method ofclaim 44, wherein said periodic index ranges from low periodic index tohigh periodic index.